When voice packets are transmitted through a communications network for providing VoIP (Voice over Internet Protocol), the regularity of intervals between the receipt of consecutive packets is inevitably disrupted. This phenomenon is known as “network jitter.” Such jitter can cause packet losses whenever a packet arrives too late and thereby misses its playback time, and these packet losses eventually degrade voice quality. In typical state-of-the-art VoIP systems, a packet buffer on the receiving end is used to alleviate this problem by adding a fixed amount of initial playback delay to compensate for the jitter. This “jitter buffer” then supplies a steady stream of voice packets into a playout mechanism, starting after the fixed delay. Thus, any amount of jitter up to the amount of the fixed delay will be accommodated with no loss of voice quality. Only packets whose delay in transmission exceeds this fixed delay will be “lost.” (For such “lost” packets, most systems employ conventional packet loss concealment techniques in an attempt to limit the resulting degradation of voice quality. Nonetheless, excessive packet loss will invariably result in significant voice quality degradation.)
In co-pending U.S. patent application Ser. No. 11/062,966, “Method And Apparatus For Handling Network Jitter In A Voice-Over IP Communications Network Using A Virtual Jitter Buffer And Time Scale Modification,” filed by M. Lee et al. on Feb. 22, 2005, and commonly assigned to the assignee of the present invention, a novel technique for use in a VoIP network which advantageously handles network jitter at the receiver without introducing additional playback latency was disclosed. In particular, the novel method disclosed therein makes use of a virtual jitter buffer and time scale modification (i.e., expansion or contraction of the voice data contained in a voice packet).
More specifically, in one embodiment disclosed therein, a method identifies a sequence of received voice packets as comprising a talk spurt, and then, starting with a virtual jitter buffer having an initial playback latency of zero, the method advantageously performs time-expansion prior to playing out the first several voice packets of the talk spurt, thereby increasing the effective latency of the jitter buffer (and therefore the amount of jitter delay handled) until a predetermined maximum effective latency is reached. (A “talk spurt” is a segment of a talker's speech preceded and followed by silence.) Then, subsequent packets are played out at their normal length (i.e., at normal speed), until the receipt of a packet indicative of the end of a talk spurt is detected, at which point the remaining (un-played out) voice packets are time-compressed as they are played out to return the effective jitter buffer latency to zero. Co-pending U.S. patent application Ser. No. 11/062,966, “Method And Apparatus For Handling Network Jitter In A Voice-Over IP Communications Network Using A Virtual Jitter Buffer And Time Scale Modification,” is hereby incorporated by reference as if fully set forth herein.
Although the above-described talk spurt management technique disclosed in co-pending U.S. patent application Ser. No. 11/062,966, “Method And Apparatus For Handling Network Jitter In A Voice-Over IP Communications Network Using A Virtual Jitter Buffer And Time Scale Modification,” identified above, advantageously provides network jitter protection for most of the packets of a given talk spurt, it is clear that at least the first several packets of a talk spurt (i.e., those that are advantageously time-expanded in order to increase the playback latency of the virtual jitter buffer) are not as protected from network jitter as the rest of the packets in the talk spurt. Thus, even with use of this talk spurt management technique, there remains some increased risk of packet loss at the beginning of each talk spurt.